This invention relates generally to Internet communications, and more particularly to Quality of Service techniques for transmitting packets from different communication sources across a packet switched network. Successful transmission of both synchronous content (e.g., voice) and asynchronous content (e.g., data) requires careful scheduling, particularly if the packet load from the asynchronous source can meet or exceed the network capacity. This creates the possibility of filling most of the network channel bandwidth with data packets, thereby undesirably delaying transmission of real-time voice conversation packets, causing delay jitters or even the loss of voice packets. If such delay jitters or losses of voice packets are not prevented, then the quality of voice content received at a destination may be severely degraded and may even cause the listener to drop the voice call. For example, when one or more users initiate data traffic during a Voice over IP call, the data will interfere with the voice packets causing poor voice quality at the listening destination. This problem is often a result of limited bandwidth in network access links such as cable and DSL that have slow upstream feed rates. While one solution to this problem is to restrict data packet throughput, excessive restrictions then may cause severe throughput loss to data transmissions, which is also not desirable.
Therefore, a solution is needed for a Voice over IP communication system that provides efficient use of network transmission links for both data and voice traffic, while at the same time establishing predictable Quality of Service for both data and voice transmissions. Prior attempts typically involved simply giving a higher priority to voice packets and/or slowing down data packet rates by inserting a gap of some fixed length between successive data packets.